The present invention relates generally to telecommunication systems that utilize digital audio/speech coders during data transmission. More particularly, the present invention relates to a telecommunication system that supports conventional waveform encoding protocols for use during relatively narrowband audio transmissions and audio coding protocols for use during relatively wideband audio transmissions over a telephone network.
Telecommunication systems such as the public switched telephone network (PSTN) and private branch exchanges (PBXs) are generally well known. The PSTN is now considered to be a digital system that is capable of carrying data at a theoretical speed of 64 kilobits per second (kbps). FIG. 1 is a general block diagram of a conventional telecommunication system 100 showing a simple end-to-end connection between a first terminal equipment device 102 and a second terminal equipment device 104. Each of the terminal equipment devices 102 and 104 may be, e.g., a standard telephone, a personal computer having telecommunication features, or a modem device.
First device 102 is operatively coupled to a first central office 106 via a local loop connection 108. Although not shown, first central office 106 is similarly coupled to a plurality of other terminal equipment devices. First central office 106 contains a first line card 110, which may include any number of components that perform call processing, switching, transmitting, receiving, and the like. First central office 106, and first line card 110 in particular, is operatively coupled to a digital telephone network 112, e.g., a PSTN, which is configured to transmit voice calls, data, and signaling information at a theoretical speed of 64 kbps.
For the general telecommunication system 100 shown in FIG. 1, signals are transmitted through network 112 to a second central office 114 that contains a second line card 116. In accordance with conventional terminology, central offices 106 and 114 are not considered to be part of network 112. Second central office 114 is coupled to second device 104 via a local loop connection 118. First and second central offices 106 and 114 include coder/decoder (codec) elements (not shown in FIG. 1) that are configured to transmit and receive data in accordance with compatible encoding techniques.
Current telephone networks are governed by international standards; such standards require voice calls to be transmitted in accordance with well-established pulse code modulation (PCM) encoding techniques, e.g., xcexc-law encoding in North America or A-law encoding in Europe. PCM encoding transmits codewords that represent analog voltage levels associated with an audio waveform. Due to the required PCM encoding techniques and the required 8 kHz symbol sampling rate, the audio signals transmitted during conventional telephone calls are limited to a bandwidth of about 4 kHz (in practical telecommunication systems, this bandwidth is reduced to about 3.5 kHz due to the use of filters, hybrids, and other functional elements by the system). While this narrow bandwidth may be suitable to provide clear and intelligible voice transmissions, the fidelity of such transmissions does not approach the audio quality associated with actual person-to-person transmissions.
In lieu of PCM codecs, digital voice/speech coders may be utilized by a telecommunication system to transmit audio signals in a different manner than the conventional PCM encoding techniques. Assuming that a suitable transmit bandwidth is available, such audio coders can provide enhanced fidelity voice transmissions by incorporating audio characteristics such as tone, pitch, resonance, and the like, into the transmitted signal. For example, by leveraging the 64 kbps capability of current telephone networks, wideband voice coders may be designed to provide high fidelity telephone calls in lieu of conventional audio calls that are governed by the PCM encoding protocols. Such high fidelity calls may be transmitted using a bandwidth that exceeds 4 kHz, e.g., 7 kHz or more.
Due to the current standards that govern telecommunication systems, audio coders may not be universally implemented in the many central offices associated with a given telecommunication system. Accordingly, an end-to-end high fidelity speech connection may not always be obtainable if either of the respective central office line cards do not utilize compatible audio coders. Even if both ends support high fidelity speech communications, there must be a mechanism by which the central offices can communicate to determine whether (and which) wideband audio coding protocols are supported. Currently, there is no efficient and practical signaling technique that enables such communication between two central offices.
A possible signaling technique may simply employ a substantial portion of the normal operating bandwidth to transmit tones or other signals at the beginning of a communication session. Although this procedure may effectively convey the necessary information between the central offices, the transmission of the signaling information may interfere with a call in progress and be noticeable to the end users. Accordingly, it would be desirable to employ signaling techniques that are not significantly perceptible to the end users.
Even if compatible audio coders are employed by both central offices associated with a given call, there may be situations where it is not desirable or necessary to enter the enhanced audio quality mode or where it may be desirable to switch back to conventional narrowband operation while operating in the enhanced audio mode. For example, the audio coding protocols may not be necessary in the context of modem connections that transmit data rather than audio information.
Therefore, it would be desirable to implement a telecommunication system that addresses the foregoing limitations of the prior art.
Accordingly, it is an advantage of the present invention that an improved telecommunication system that utilizes audio coders is provided.
Another advantage of the present invention is that it may be incorporated into an existing telecommunication system while maintaining backward compatibility with conventional PCM encoding protocols.
Another advantage is that the present invention provides a signaling mechanism by which two central offices associated with a telephone network can communicate to determine whether they are compatible with wideband audio coding protocols.
A further advantage is that the present invention provides a signaling procedure for use in connection with the wideband audio mode, where the signaling procedure is performed in a manner that is minimally perceptible to the end users.
The above and other advantages of the present invention may be carried out in one form by a telecommunication method for providing an enhanced audio quality communication session over a digital telephone network having first and second codecs operatively coupled thereto, where the first and second codecs are associated with first and second terminal equipment devices, respectively. The method first establishes a communication session between the first and second codecs, where the communication session is governed by a first data communication protocol under which audio transmissions are limited to a first bandwidth. Next, the method transmits, during the same communication session, an in-band signaling sequence from the first codec to the second codec to determine whether the second codec is compatible with a second data communication protocol under which audio transmissions are limited to a second bandwidth that exceeds the first bandwidth.